A current option in voice telephony services is to convert standard voice transmissions to data packets which can be transmitted over the Internet, e.g. Voice over Internet Protocol (VoIP) and the like. Such VoIP services can be connected to an existing wireline Public Switched Telephone Network (PSTN) in order to provide compatibility with existing equipment.
As is known in the art, PSTN system uses telephone numbers (e.g. 1-888-555-2222) to direct telephone calls, while VoIP uses a Domain Name System (DNS) (e.g. sam@address.com) which is converted to a numeric IP address (e.g. 111.222.333.444) to direct the data packets. Alternatively, the VoIP system can address data packets using a Session Initiation Protocol Uniform Resource Identifier (SIP URI) as is known in the art (e.g. sip:sam@111.222.333.444, sip:sam@voip.example.com, sip: 18885552222@voip.example.com, and the like). When traversing from the PSTN network to the VoIP network, it is necessary to translate the PSTN telephone number into one of these address forms usable by the VoIP data network.
To accommodate the change in addressing modes, the Internet Engineering Task Force (IETF) introduced the E.164 Number Mapping (ENUM) system (RFC 3761) to transform E.164 numbers (The International Public Telecommunication Numbering Plan, ITU-T Recommendation E.164) into SIP URI addresses or/and domain names. The ENUM system then uses DNS delegation (RFC 1034) through Name Server (NS) records and Naming Authority Pointer (NAPTR) records (RFC 3403) to look up the corresponding sip URI or/and domain name for the given E.164 number. A call originating in a PSTN network uses the E.164 number and PSTN routing decisions to route the call into the VoIP network. An ENUM lookup in a DNS server is used to find the ingress point into the VoIP network, as well as the VoIP URI of the callee.
However, a problem arises where the call is to egress back into a PSTN network domain for termination at a callee therein. E164 to SIP ENUM lookups do not guarantee the preservation of the original called number. Once in the VoIP network, the original dialed E.164 number may get lost in the process either due to service invocations, which may cause the SIP URI to change along the VoIP path, or due to VoIP entities not keeping track of it. In other words, the VoIP network may not know the routing direction back to a PSTN domain.
One solution to the egress problem is to define a static route to a specific media gateway based on trunk groups. Another solution is to have the session routed to a Signaling gateway which then specifies a media gateway to route to the appropriate Media gateway to egress the VOIP network to the PSTN network. In both cases, egress from the VOIP network to a PSTN network is accomplished with a complicated set of routes and rules that are statically provisioned into the routing entity, such as a Serving-Call Session Control Function (S-CSCF). Another solution is to add the original called number to a SIP header that is non-standard. However, even in this case the header not be preserved across various SIP realms due to security policies that prevent the passing on of “spurious” headers.
Therefore, it is desired to provide a technique for routing a call originating and terminating in a PSTN system through a VoIP communication network. It would also be of benefit to provide such service utilizing an existing NAPTR record format instead of utilizing specialized call control functions.